Re: [Asterisk-Users] sip to sip calls thru asterisk

Thu, 26 Aug 2004 17:11:18 -0700

You may need a stun server.  I setup my own and it appears to be working just fine.
 
To * both sip clients appear to be on the same IP and that confuses *.  STUN should clear up some of that confusion.
 
I don't know enough about STUN to know if it needs to be on a public IP vs a private ip however.
 
Lyle
 
----- Original Message -----
From: Gary Carr
Sent: Tuesday, August 24, 2004 1:29 PM
Subject: [Asterisk-Users] sip to sip calls thru asterisk

I have a test box setup and I can make outbound calls on the PSTN thru the diguim card, however I can not make a sip user to sip user call by dialing the extensions. I am getting the following error.
 
-- Called cisco7960
    -- Got SIP response 482 "Loop Detected" back from 208.218.14.123
  == No one is available to answer at this time
 
 
 
CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port     Status
 
cisco7960/5052   208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)
garycarr/5011    208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)
 
 
sip.conf statements
 
 
[cisco7960]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5052
callerid="Cisco 7960"
context=local
 
[garycarr]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5011
callerid="Gary Carr"
context=local
 
extensions.conf statements
 
exten => 5011,1,dial(SIP/garycarr,20,tr)
exten => 5052,1,dial(SIP/cisco7960,20,tr)
 
Is this a possible nat issue? I can make a good call from behind the firewall doing sip to pstn so it seems 2 way traffic thru the firewall is working.
 
 
I am still sifting thru the sip debug info but anyone has any ideas that would be great.
 
 
Gary
 

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